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| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/modules/audio_processing/test/echo_canceller_test_tools.h" | |
| 12 | |
| 13 namespace webrtc { | |
| 14 | |
| 15 void RandomizeSampleVector(Random* random_generator, rtc::ArrayView<float> v) { | |
| 16 for (auto& v_k : v) { | |
| 17 v_k = 2 * 32767.f * random_generator->Rand<float>() - 32767.f; | |
| 18 } | |
| 19 } | |
| 20 | |
| 21 // Produces a delayed signal copy of x. | |
|
hlundin-webrtc
2017/01/23 19:48:28
Repeated comment. Drop this.
peah-webrtc
2017/01/23 21:38:22
Done.
| |
| 22 template <typename T> | |
| 23 void DelayBuffer<T>::Delay(rtc::ArrayView<const T> x, | |
| 24 rtc::ArrayView<T> x_delayed) { | |
| 25 RTC_DCHECK_EQ(x.size(), x_delayed.size()); | |
| 26 if (buffer_.size() == 0) { | |
| 27 std::copy(std::begin(x), std::end(x), std::begin(x_delayed)); | |
| 28 } else { | |
| 29 for (size_t k = 0; k < x.size(); ++k) { | |
| 30 x_delayed[k] = buffer_[next_insert_index_]; | |
| 31 buffer_[next_insert_index_] = x[k]; | |
| 32 next_insert_index_ = (next_insert_index_ + 1) % buffer_.size(); | |
| 33 } | |
| 34 } | |
| 35 } | |
| 36 | |
| 37 template class DelayBuffer<float>; | |
| 38 template class DelayBuffer<int>; | |
| 39 } // namespace webrtc | |
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